Signaling may often be used in the public switched telephone network (PSTN) to set up and terminate circuits, sessions, and so on. For example, one of the signaling protocols for the PSTN is called Common Channel Signaling System 7 (SS7). In the PSTN, existing signaling protocols may provide support for basic call setup and tear down, billing, wireless services and roaming, local number portability, toll-free calling services, caller ID, three-way calling, call forwarding enhanced features, etc. However, these existing signaling protocols provide limited support. For example, with respect to caller ID, there is no guarantee that caller ID information will traverse end to end. Another issue is that the caller ID is not unique to a user if the call originates from a private branch exchange (PBX) extension, or the like. For example, when an Acme Corporation employee places a call, it may identify “Acme” rather than the employee.
Unlike PSTN signaling protocols, session initiation protocol (SIP) is not limited to communication during set signaling phases. SIP is an application-layer control protocol that computer systems can use to discover one another and to establish, modify, and terminate multimedia sessions. For example, SIP is one of the key protocols used in implementing Voice over IP technology. Implementing Voice over IP technology typically involves converting voice information to digital form and sending it in discrete packets rather than in the traditional circuit-committed protocols of the PSTN. SIP is also associated with implementing features such as instant messaging and other “real time” communication techniques. For example, an instant messaging service allows participants to send messages and have them received within a second or two by the other participants in the conversation. The receiving participants can then send responsive messages to the other participants in a similar manner.
SIP is an Internet proposed standard. Its specification, “RFC 3261,” is available at <http://www.ietf.org/rfc/rfc3261.txt>. A specification for extensions to SIP relating to event notifications, “RFC 3265,” is available at <http://www.ietf.org/rfc/rfc3265.txt>. A SIP network comprises entities that can participate in a dialog as a client, server, or both. SIP supports four types of entities: user agent, proxy server, redirect server, and registrar. User agents initiate and terminate sessions by exchanging messages with other SIP entities. A user agent can be a user agent client (“UAC”), which is a device that initiates SIP requests, or a user agent server (“UAS”), which is a device that receives SIP requests and responds to such requests. As examples, “IP-telephones,” personal digital assistants, and any other type of computing device may be user agents. A device can be a UAC in one dialog and a UAS in another, or may change roles during the dialog. A proxy server is an entity that acts as a server to clients and a client to servers. In so doing, proxy servers intercept, interpret, or forward messages between UACs and UASs. A redirect server accepts a SIP request and generates a response directing the UAC that sent the request to contact an alternate network resource. A registrar is a server that accepts registration information from user agents and informs a location service of the received registration information.
SIP supports two message types: requests, which are sent from a UAC to a UAS, and responses, which are sent from a UAS to a UAC when responding to a request. A SIP message is composed of three parts. The first part of a SIP message is a “request line,” which includes fields to indicate a message method (e.g., INVITE) and a Request URI that identifies the user or service to which the request is being directed. The second part of a SIP message comprises headers whose values are represented as name-value pairs. The third part of a SIP message is the message's body, which is used to describe the session to be initiated or contain data that relates to the session. Message bodies may appear in requests or responses.